mirror of
https://github.com/hrydgard/ppsspp.git
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The ffmpeg submodule held prebuilt libraries for every platform next to the sources (about 1.5 GB checked out), and nothing uses those any more. Only the needed sources are copied in, from the submodule's commit, along with every header and template they include. The path is new, so checking out commits on either side of this works. It's also cut down to what PPSSPP uses: the H.264, AAC, MP3, ATRAC3 and ATRAC3+ decoders, the H.264 parser, the demuxers as before, and the FFV1 encoder, AVI muxer and file protocol for AVI dumping. config.h takes its CONFIG_ values from FFmpeg 3.0.2's own configure run with just those enabled. The raw H.264 demuxer stays even though nothing opens raw H.264: probing PSMF data uses it to identify the video stream's codec, which otherwise comes out as none. On top of that: - The files nothing links (configure would build them, but they're never pulled out of the static library, going by the GCC link map) are left out: mostly libavutil's crypto and hashing and the alternative DCT/IDCT/FFT versions. - libswresample isn't built, now that nothing uses it. - The H.264 decoder only does 8-bit video, which is all the PSP has. The SPS parser rejects any other bit depth, and the DSP files no longer build their 9- to 16-bit variants, which the linker kept because the init functions pick one by bit depth at run time. That's 160 files to compile instead of 277, and 391 of FFmpeg's vendored, about 5.8 MB. The README in ext/ffmpeg has what was done, the PPSSPP patches in the copy, how to add a component, and the tradeoffs. Co-Authored-By: Claude Opus 5.5 (1M context) <noreply@anthropic.com>
253 lines
8.7 KiB
C
253 lines
8.7 KiB
C
/*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "common.h"
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#include "samplefmt.h"
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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typedef struct SampleFmtInfo {
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char name[8];
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int bits;
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int planar;
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enum AVSampleFormat altform; ///< planar<->packed alternative form
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} SampleFmtInfo;
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/** this table gives more information about formats */
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static const SampleFmtInfo sample_fmt_info[AV_SAMPLE_FMT_NB] = {
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[AV_SAMPLE_FMT_U8] = { .name = "u8", .bits = 8, .planar = 0, .altform = AV_SAMPLE_FMT_U8P },
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[AV_SAMPLE_FMT_S16] = { .name = "s16", .bits = 16, .planar = 0, .altform = AV_SAMPLE_FMT_S16P },
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[AV_SAMPLE_FMT_S32] = { .name = "s32", .bits = 32, .planar = 0, .altform = AV_SAMPLE_FMT_S32P },
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[AV_SAMPLE_FMT_FLT] = { .name = "flt", .bits = 32, .planar = 0, .altform = AV_SAMPLE_FMT_FLTP },
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[AV_SAMPLE_FMT_DBL] = { .name = "dbl", .bits = 64, .planar = 0, .altform = AV_SAMPLE_FMT_DBLP },
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[AV_SAMPLE_FMT_U8P] = { .name = "u8p", .bits = 8, .planar = 1, .altform = AV_SAMPLE_FMT_U8 },
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[AV_SAMPLE_FMT_S16P] = { .name = "s16p", .bits = 16, .planar = 1, .altform = AV_SAMPLE_FMT_S16 },
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[AV_SAMPLE_FMT_S32P] = { .name = "s32p", .bits = 32, .planar = 1, .altform = AV_SAMPLE_FMT_S32 },
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[AV_SAMPLE_FMT_FLTP] = { .name = "fltp", .bits = 32, .planar = 1, .altform = AV_SAMPLE_FMT_FLT },
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[AV_SAMPLE_FMT_DBLP] = { .name = "dblp", .bits = 64, .planar = 1, .altform = AV_SAMPLE_FMT_DBL },
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};
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const char *av_get_sample_fmt_name(enum AVSampleFormat sample_fmt)
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{
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if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
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return NULL;
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return sample_fmt_info[sample_fmt].name;
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}
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enum AVSampleFormat av_get_sample_fmt(const char *name)
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{
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int i;
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for (i = 0; i < AV_SAMPLE_FMT_NB; i++)
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if (!strcmp(sample_fmt_info[i].name, name))
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return i;
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return AV_SAMPLE_FMT_NONE;
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}
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enum AVSampleFormat av_get_alt_sample_fmt(enum AVSampleFormat sample_fmt, int planar)
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{
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if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
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return AV_SAMPLE_FMT_NONE;
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if (sample_fmt_info[sample_fmt].planar == planar)
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return sample_fmt;
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return sample_fmt_info[sample_fmt].altform;
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}
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enum AVSampleFormat av_get_packed_sample_fmt(enum AVSampleFormat sample_fmt)
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{
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if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
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return AV_SAMPLE_FMT_NONE;
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if (sample_fmt_info[sample_fmt].planar)
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return sample_fmt_info[sample_fmt].altform;
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return sample_fmt;
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}
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enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt)
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{
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if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
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return AV_SAMPLE_FMT_NONE;
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if (sample_fmt_info[sample_fmt].planar)
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return sample_fmt;
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return sample_fmt_info[sample_fmt].altform;
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}
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char *av_get_sample_fmt_string (char *buf, int buf_size, enum AVSampleFormat sample_fmt)
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{
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/* print header */
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if (sample_fmt < 0)
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snprintf(buf, buf_size, "name " " depth");
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else if (sample_fmt < AV_SAMPLE_FMT_NB) {
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SampleFmtInfo info = sample_fmt_info[sample_fmt];
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snprintf (buf, buf_size, "%-6s" " %2d ", info.name, info.bits);
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}
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return buf;
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}
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int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
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{
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return sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB ?
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0 : sample_fmt_info[sample_fmt].bits >> 3;
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}
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int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
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{
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if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
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return 0;
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return sample_fmt_info[sample_fmt].planar;
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}
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int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples,
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enum AVSampleFormat sample_fmt, int align)
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{
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int line_size;
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int sample_size = av_get_bytes_per_sample(sample_fmt);
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int planar = av_sample_fmt_is_planar(sample_fmt);
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/* validate parameter ranges */
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if (!sample_size || nb_samples <= 0 || nb_channels <= 0)
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return AVERROR(EINVAL);
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/* auto-select alignment if not specified */
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if (!align) {
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if (nb_samples > INT_MAX - 31)
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return AVERROR(EINVAL);
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align = 1;
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nb_samples = FFALIGN(nb_samples, 32);
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}
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/* check for integer overflow */
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if (nb_channels > INT_MAX / align ||
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(int64_t)nb_channels * nb_samples > (INT_MAX - (align * nb_channels)) / sample_size)
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return AVERROR(EINVAL);
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line_size = planar ? FFALIGN(nb_samples * sample_size, align) :
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FFALIGN(nb_samples * sample_size * nb_channels, align);
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if (linesize)
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*linesize = line_size;
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return planar ? line_size * nb_channels : line_size;
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}
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int av_samples_fill_arrays(uint8_t **audio_data, int *linesize,
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const uint8_t *buf, int nb_channels, int nb_samples,
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enum AVSampleFormat sample_fmt, int align)
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{
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int ch, planar, buf_size, line_size;
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planar = av_sample_fmt_is_planar(sample_fmt);
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buf_size = av_samples_get_buffer_size(&line_size, nb_channels, nb_samples,
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sample_fmt, align);
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if (buf_size < 0)
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return buf_size;
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audio_data[0] = (uint8_t *)buf;
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for (ch = 1; planar && ch < nb_channels; ch++)
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audio_data[ch] = audio_data[ch-1] + line_size;
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if (linesize)
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*linesize = line_size;
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return buf_size;
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}
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int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels,
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int nb_samples, enum AVSampleFormat sample_fmt, int align)
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{
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uint8_t *buf;
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int size = av_samples_get_buffer_size(NULL, nb_channels, nb_samples,
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sample_fmt, align);
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if (size < 0)
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return size;
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buf = av_malloc(size);
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if (!buf)
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return AVERROR(ENOMEM);
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size = av_samples_fill_arrays(audio_data, linesize, buf, nb_channels,
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nb_samples, sample_fmt, align);
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if (size < 0) {
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av_free(buf);
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return size;
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}
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av_samples_set_silence(audio_data, 0, nb_samples, nb_channels, sample_fmt);
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return size;
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}
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int av_samples_alloc_array_and_samples(uint8_t ***audio_data, int *linesize, int nb_channels,
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int nb_samples, enum AVSampleFormat sample_fmt, int align)
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{
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int ret, nb_planes = av_sample_fmt_is_planar(sample_fmt) ? nb_channels : 1;
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*audio_data = av_calloc(nb_planes, sizeof(**audio_data));
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if (!*audio_data)
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return AVERROR(ENOMEM);
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ret = av_samples_alloc(*audio_data, linesize, nb_channels,
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nb_samples, sample_fmt, align);
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if (ret < 0)
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av_freep(audio_data);
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return ret;
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}
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int av_samples_copy(uint8_t **dst, uint8_t * const *src, int dst_offset,
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int src_offset, int nb_samples, int nb_channels,
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enum AVSampleFormat sample_fmt)
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{
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int planar = av_sample_fmt_is_planar(sample_fmt);
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int planes = planar ? nb_channels : 1;
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int block_align = av_get_bytes_per_sample(sample_fmt) * (planar ? 1 : nb_channels);
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int data_size = nb_samples * block_align;
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int i;
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dst_offset *= block_align;
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src_offset *= block_align;
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if((dst[0] < src[0] ? src[0] - dst[0] : dst[0] - src[0]) >= data_size) {
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for (i = 0; i < planes; i++)
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memcpy(dst[i] + dst_offset, src[i] + src_offset, data_size);
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} else {
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for (i = 0; i < planes; i++)
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memmove(dst[i] + dst_offset, src[i] + src_offset, data_size);
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}
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return 0;
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}
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int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples,
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int nb_channels, enum AVSampleFormat sample_fmt)
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{
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int planar = av_sample_fmt_is_planar(sample_fmt);
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int planes = planar ? nb_channels : 1;
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int block_align = av_get_bytes_per_sample(sample_fmt) * (planar ? 1 : nb_channels);
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int data_size = nb_samples * block_align;
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int fill_char = (sample_fmt == AV_SAMPLE_FMT_U8 ||
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sample_fmt == AV_SAMPLE_FMT_U8P) ? 0x80 : 0x00;
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int i;
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offset *= block_align;
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for (i = 0; i < planes; i++)
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memset(audio_data[i] + offset, fill_char, data_size);
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return 0;
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}
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